Hi all, we are currently in the middle of testing wtih BT for their new VOIP platform and things have generally gone quite well but we have one test that we really need to pass. A incoming call from BT needs to be forwarded back out to another BT number. When this is done from PSTN phones everything works OK but when BT use SIP phones they get one of two scenarios.
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Extra invite packets during the call. This can be fixed by setting CanReInvite = NO but then we get…
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No Audio. If we stop the extra invites then we lose the audio on the forwarded call, this audio loss i believe is both ways.
Anyone experienced anything similair to this? Have any ideas how to fix the problem?
Any help would be appreciated.
Kev
I advise you test it with x-lite, if it is ok, then you can contact your sip pohone provider. As I know, Jin301 worked like this.
Leon
telecomchinasourcing.com
leon, thanks for the reply.
We are using x-lite for our tests but BT use their own sip phones. The problem seems to be because Asterisk is trying to keep itself involved in the calls, the rtp invites are sent from asterisk rather than from the two end points effectively creating a conference call rather than a true forwarded call.
This is with canreinvite=yes, if you set this to no then there are no reinvites but there’s no audio either. The whole issue kind of comes around because asterisk is designed to be more of an “end point”. We’ve found a SIP proxy called “SIP Express Router” that seems to do what we want for now. It would be nice if we could use asterisk instead though so if anyone has a better solution please let me know.
Cheers
Kev